I found these quotes from Elias Gwinn, an engineer at Benchmark Media:

ALL headphones have non-linear mechanical impedances (that is, the mass and shape of a speaker will resonate more at certain frequencies and much less at other frequencies). This means the physical build of the headphones (as well as other physical impedances, like your head and ears!) will try to override the electrical system (amplifier and speaker coil).

To create low-distortion headphone response, one must consider 'damping factor'. A high damping factor will control the response of the speaker, thus preventing the physical impedances from dictating frequency response. Damping factor is the ratio of speaker (load) impedance to amplifier (source) impedance. In other words, the best damping factor will result from a low source impedance. Again, the source impedance from the HPA2 is less then 0.01 ohms...as low as gets!!

Balanced headphone amps will double the source impedance of an unbalanced headphone amp. No matter how low the impedance of a balanced headphone amp, it could be half that much if it was unbalanced. This is one reason balanced headphone amps are not a good idea. (It should also be noted that the balanced output of the DAC1 / USB / PRE is 60 ohms or greater, depending on the attenuator settings).

Not only will the source impedance double with balanced headphone amplifiers, but the total distortion and noise of the amplifier will double as well!! Every output device (opamp, transistor, tube) creates some distortion and some noise. If there are two opamps or transistors or tubes driving each headphone speaker, twice as much distortion and noise will be added!!

The result of balanced headphones is less damping factor, more distortion, and more noise. Also, balanced headphones configurations offer no real benefits, to boot.

Feel free to use the XLR outputs of the DAC1 / USB / PRE for balanced headphone outputs (as mentioned above, the DAC1 USB and DAC1 PRE will do better then the DAC1 at this task, because of the 4562's). It won't damage anything to operate in this configuration. But, for the reasons above, I don't recommend it.

We've been getting a lot of questions lately about balanced headphones. We are interested in the debate, but I can't say we agree with any technical explanations about the benefits of the set up.

So far, there are 4 major points mentioned so far (that I have heard, at least):

1. Unshared common conductor reduces crosstalk

2. Two amps (per channel) increases slew-rate

3. Two amps provide better damping

4. Balanced cabling provides better common-mode rejection

If I may, I'd like to add my thoughts on these points:

1. Most headphones (at least those of decent quality) do not share a common conductor through the length of the cable (as opposed to what was said in 6 Moons). Most headphones have a separate wire from each negative terminal that remain isolated through the length of the cable. In other words, most headphone cables are effectively balanced inherently. If they were sharing a common through the length of the cable, the impedance of the cable may cause some of the signal to show up on opposing channels. However, they are not connected until the plug, and therefore have a minimal impedance to ground.

2. Any headphone amp that is struggling with slew-rate is a poorly designed headphone amplifier. The HPA2 headphone amplifier on the DAC1 has a bandwidth of 55 kHz, and it doesn't even approach any slew-rate limitations even at those high frequencies.

3. Two amps provide WORSE damping. This is why power amplifiers run better in normal mode vs. bridged mode. A balanced (dual-active) headphone amplifier is exactly analogous to a bridged amplifier driving one speaker. The only advantage is increased power, but it comes at an expense of increased distortion, decreased damping, and altered frequency response. This is common knowledge for bridged amplifiers.

4. Headphones don't need any help with common-mode rejection because they inherently will not respond to common mode signal. If, for example, you apply a signal to both terminals of a speaker, it will not move at all. A speaker only responds to differential voltages.

5. There is another cost incurred by dual-active headphone amps that is not addressed. Headphone amps should have as low of a source impedance as possible. If you are using two amps to drive a channel, you are doubling the source impedance. This will cause the headphones to suffer in frequency response, distortion, and ringing.

Please continue the great discussions. It is important to resolve these debates so that product manufactures can respond to provide the best audio solutions possible.

While Benchmark's statement that having two amplifiers drive the headphones (per channel, in balanced mode) doubles its effective output impedance is true, for any AMB headphone amp, the intrinsic output impedance is so low, that the output impedance difference between balanced and unbalanced is effectively nil. A miniscule fraction of one ohm, when doubled, is still a miniscule fraction of one ohm. When driving a headphone load of tens of ohms to hundreds of ohms in impedance, the damping factor is huge either way.

Mr. Gwinn's remark about common-mode rejection is also misleading. When he said that feeding an identical signal to both sides of the transducer would result in no sound, he is is precisely describing the benefit of common mode rejection! This is why XLR balanced cabling is most commonly used in pro settings (concert/stage, recording studio, radio/broadcast, sound reinforcement, etc., where noise immunity in long cable runs is of paramount importance). Noise interference will affect both phases of the balanced wiring, thus the difference between them is zero (i.e., canceled).

That said, I do agree that CMR is not an important factor in headphone (or speaker) wiring, because the output impedance of the amp is so low that noise interference through the wiring is not going to be a problem. If the source is balanced, it makes sense to preserve the balanced mode of operation through the entire chain (i.e., with balanced amplification and balanced headphones too). Benchmark's own DAC1s have balanced XLR line outputs for driving balanced gear. Those outputs are not good for driving headphones directly (they don't have the cojones to drive low-Z headphones satisfactorily, and their high-ish output impedance is a manifestation of that), but they could be used to drive a balanced amp very well indeed.

The statement about distortion and noise being additive in a balanced setup is also misleading. As "seen" by the headphone load, if the hot and cold side amplifiers are identical, then the distortion products cancel each other.

With all due respect to Benchmark, I own a DAC1 and think it's a very good product, but I believe Mr. Gwinn's remark is dumbed-down to appease an audience at head-fi who are not amp designers or engineers, because there is quite a bit more to the case for balanced amplification than what he mentioned. Perhaps he is justifying Benchmark's lack of a balanced headphone output.

I had posted in the past on other forums about how supply rail currents in a balanced class A amp (as well as the 3-channel active ground configuration) cancel out, resulting in a net zero current flow. This, along with the fact that the headphone load is not referenced to ground, makes the signal as seen by the load to be totally immune to distortion products caused by ground pollution (which occurs in conventional 2-channel passive-ground amps). Such ground pollution occurs when the headphone transducer's return current flows through the ground wiring and ESR of the power supply bulk caps, inducing voltage wiggles in the ground. Due to the fact that real world loads are reactive, the return currents are phase-shifted relative to the amp's output voltage swing. So the "ground voltage wiggle" is also phase-shifted to the original signal, and when you sum the two, you get distortion.

As for bandwidth and slew rate, have a look at the specifications of any AMB amplifier, and you'll see how Benchmark's 55KHz compare.

We could generalize as Mr. Gwinn did, and it would not prove one thing or another. However, one look at the actual measured results of the group build balanced β22 (For pics, visit the β22 website gallery section, and click on "group-build"), will show you that it gives up nothing in performance compared to any amp, balanced or otherwise:

Note that the actual performance of the amp is better than these figures and graphs because the results approach the limits of the measurement sound card (E-MU 0404 USB). For example, the frequency response rolloff of the amp is actually flat from 0Hz to 2.5MHz -3dB, much of the distortion products are of the sound card itself, and the rising stereo crosstalk graph at high frequencies is also an artifact of the input attenuator pots on the E-MU 0404 USB.

Ti,Thanks for the (once again) very imformative analysis and comparison. I also appreciate the graphs, since I had not seen these published before. I also completely agree with your statements regarding the balanced chain of amplification. Since I worked with the recording & mastering industry I noted that these professionals only used balanced signal lines for the very reasons you have stated. Grounds are always a challenge in the recording studios as musicians would have various pieces of equipment that would have poor grounds and/or signal chains. Using balanced connections eliminates many problems, if done correctly.

I have not worked in a studio, but its my understanding that 95% of what pros want from balanced is CMR on the CABLES. nothing to do with sound wrt to 'higher slew rate' or reduced distortion. output-z is also something most studio guys (I think) care nothing about. this is a DESIGNER issue, not anything studio engineers ('users') can do anything about other than selecting which box to purchase.

in studios and on stage, you have LONG cable runs and lots of ac around it. at home, you never have 'long' cable runs and your ac is usually not as bad as stage or studio. I do not believe in using balanced mode cabling for home use in 95% of the cases. its simply not needed and, yes, it does add extra 'stuff' in your signal path. that should bother the purists, right?

differential mode analog signalling is a great idea; but that's not the same exact thing as having 2 amps, one in phase and one out of phase, pumping signal down a wire. if the wire is a few feet long, I personally think you will never need balanced cabling.

and there are better ways to solve ground loops than to have 'double the amplifiers' in your signal chain.

so my take is: low level signals on long cables (microphones on a stage): YES! use balanced. high level signals (phones or speakers) do not 'need' balanced mode and it really is arguable if double the electronics is useful for a few feet of 'cable drive' that you have to deal with.

also, in balanced mode, you have to be VERY careful in matching the 2 'sides' of the wire very very well. its not easy to do and if you are off by even a little, there goes your common mode rejection that you fought so hard to have! both amps have to track exactly (and I mean EXACTLY) and so does the power supply. in practice, getting a very high CMRR is not easy.

I do agree with the ground pollution concept but that's NOT balanced mode; that's a separate thing entirely. you can have a 3ch amp and not be in balanced mode. I think that adds value more than the 2nd amp per channel does.

I have always been curious about the "slew rate advantage" of balanced configurations... Wouldn't slew rate advantages be limited by the audible frequency range? Are there advantages to having a slew rate that exceeds this?

Iniamyen wrote:I have always been curious about the "slew rate advantage" of balanced configurations... Wouldn't slew rate advantages be limited by the audible frequency range? Are there advantages to having a slew rate that exceeds this?

Slew rate is defined as the maximum voltage the output of an amplifier could swing in a specific unit of time, usually specified as V/uS. It is supposed to measure the speed of an amp. When balanced, the hot and cold side swing voltage at the same time differentially, resulting in doubling of V. Thus the slew rate also doubles. This is purely due to doubling of output voltage swing, not because the amplifier is actually faster (assuming an apple-to-apples comparison, using the same amplifier design in both balanced and unbalanced design).

To measure slew rate you need a function generator that produces a fast-rising and falling square waves. Assuming that the function generator produces much faster square wave edges than the amplifier under test, with an oscilloscope you should see the output voltage rise and fall at the edges of the square wave in a ramp-like manner (with expanded horizontal time scale). Then, measure the amount of voltage change for the square wave to go from 10% of the swing to 90%, divided by the amount of time it takes to do so (also known as "rise time").

Note that an ideal square wave has infinite slew rate and zero rise time. Thus, for slew rate testing we're only interested in the edges of the square wave, not the period of time between cycles of the square wave, so the result should be the same whether you use a 1KHz or 100KHz square wave. The latter only makes the output change states more often in a given time period. To make a long story short, yes, the slew rate doubling of balanced amplification happens at any frequency the amplifier is capable of amplifying signal.

Here is a Tek TDS2014 scope shot of the unbalanced β22 swinging 43.2Vpp square wave at 100KHz. The top trace is the output from the function generator and the bottom trace is the output from the amplifier.

Note that the voltage swing and the rise time and fall time are calculated automatically by the scope and displayed on the right. Using 10% to 90% the voltage change would be 34.6Vpp, divided by the rise time of 175nS, we get a slew rate of 198V/uS. A balanced β22 will swing double that voltage in the same amount of time, thus the slew rate is 396V/uS.

Commentary: high slew rate is a good thing from an engineering point of view (like motherhood and apple pie). What's important is that the amplifier be much faster than any real world signal it's likely to be required to amplify -- i.e., music. With β22-level of slew rate performance it's never going to be be taxed by musical signals. It's a bit like driving a Ferrari on normal streets - you'll never approach its true performance envelope.

linux-works wrote:side question: does slew rate ever vary with frequency on a given amp? maybe some designs 'do better' at some freqs or ranges than others? or is slew rate always a static kind of property of the amp?

No, slew rate is a measure of the speed at which a voltage could rise and fall at the edge of the square wave, not of the frequency (cyclic time) of the square wave itself.